similar to: 1.6.0-rc6 - SIP hold logic broken?

Hi, I was switching from zaptel to dahdi and got latest SVN from everything. Compiling works fine. kernel module dahdi_dummy is loaded. /dev/dahdi/pseudo exists Trying to go into a meetme does not work: [Aug 11 14:04:45] -- Executing [8001 at client_int_sgmobile:1] MeetMe("SCCP/6000-00000001", "444|dcIM") in new stack [Aug 11 14:04:45] WARNING[4184]: app_meetme.c:775

The Asterisk.org development team is proud to announce the releases of Asterisk 1.4.22 and 1.6.0. ================================================================= === Asterisk 1.4.22 ============================================= ================================================================= Asterisk 1.4.22 includes a large number of bug fixes for the 1.4 release series of Asterisk. 1.4.22

The Asterisk.org development team is proud to announce the releases of Asterisk 1.4.22 and 1.6.0. ================================================================= === Asterisk 1.4.22 ============================================= ================================================================= Asterisk 1.4.22 includes a large number of bug fixes for the 1.4 release series of Asterisk. 1.4.22

Hi, I have a problem with Cisco 7906G and SIP protocol use with Asterisk 1.2.26. I have uploaded in my tftp server the firmware 'cmterm-7911_7906-sip.8-0-4SR1' that use 'SIP11.8-0-4SR1S.loads' and in SEPmacaddress.cnf.xml I have: <loadInformation>SIP11.8-0-4SR1S</loadInformation> ..but in tftp log server I have: Oct 07 11:56:22 asterisk1.local

That does not make too much sense to me... Configuration should be ok... [Aug 8 23:30:13] SSL certificate ok [Aug 8 23:30:13] == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) [Aug 8 23:30:13] WARNING[23835]: tcptls.c:463 ast_make_file_from_fd: FILE * open failed! Terve, Stefan -- Last words of a stormchaser: "Where is that rotation on the radar?!"

Hi, when I still had ISDN, I was using Hangup(causecode) to send e.g. "Wrong number" to unwelcome callers. Meanwhile, I am only using SIP providers (no PSTN lines any more) and I would like to do similar, i.e. send specific SIP headers. Besides "wrong number", I would especially like to send 302 temp moved with a specified address to deflect certain calls. Is there any way to

Hi, I have a SIP request which comes from an Asterisk and which has to re-enter in the same Asterisk (during the same session), but during the second passage in Asterisk, it send me a 482 Loop Detected. So is it a bug or Asterisk control the session and considere it as a loop ? If it is not a bug, how could I resolve this problem ? Thanks Regards -- R?mi Druilhe

Hi i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling client so i can make VOIP calls thru that phone. Aslo that Phone easly able to register with Asterisk Pbx to recive inter-office calls. i try to search from web & also from Nokia site but they only mention this features as "VOIP call from wifi" they mentioed only this much info. they not mentioed info about

>> I've tried specifying an output file as well, per the script's command line options, >> but the output file is 0 bytes.? Does anyone have any suggestions?? I *think* I'm >> using it the way it's intended to be used, but maybe I'm not?! >> -Dave > > Hi! > Maybe the key you tried was not used to encrypt the file? > Aki Aki,

I have this in sip show history for a particular channel marked as dead (should be removed) in sip show channels: 1. TxReqRel INVITE / 102 INVITE 2. Rx SIP/2.0 / 102 INVITE 3. CancelDestroy 4. Rx SIP/2.0 / 102 INVITE 5. CancelDestroy 6. Unhold SIP/2.0 7. Rx SIP/2.0 / 102 INVITE 8. CancelDestroy 9. Unhold SIP/2.0 10. Rx

Hello, I think I met a case similar to the one solved by [1] . Quoting this case : * res_pjsip: Handle deferred SDP hold/unhold properly. Some SIP devices indicate hold/unhold using deferred SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk

I was browsing through the SAMBA workgroups defined here: E51-CAE, E51-WAAS, etc... everything looked fine (well, with the exception of subnet browsing) until I displayed the properties of the machines in the workgroups. Select a SAMBA host, right-click on the hostname and select 'Properties' from an NT4 machine displays the domain as "E51"; everything from the dash onward

Not to stir the pot, but I notice my email address has recently been harvested from this list for spamming purposes. This email address is unique and not used for anything else. I'd distinguish this from spam sent to the mailing list itself, which is obviously different. Is there anything further that could be done to prevent this? -- Dave

Through the AMI how can I tell if a call is on hold or not? I am using 1.4.X Thanks, jerry

I got a very strange problem with call-hold function. For calls that come in from PSTN and route to a SIP extension. If I put the call on hold, I cannot unhold the call after. The caller would be left with hold music forever. A warning message would be shown on the console usually a few seconds after putting the call on hold: WARNING[17428]: chan_sip.c:686 retrans_pkt: Maximum retries

libswfdec/swfdec_as_context.c | 4 - test/trace/loadvars-5.swf |binary test/trace/loadvars-6.swf |binary test/trace/loadvars-6.swf.trace | 50 ++++++++----- test/trace/loadvars-7.swf |binary test/trace/loadvars-7.swf.trace | 50 ++++++++----- test/trace/loadvars.as | 4 - test/trace/loadvars.txt | 3 test/trace/propflags-5.swf

Using X-Lite FWD soft phone, I can register, get to the 'demo' menu extension, but that's it. Audio starts, then after a few seconds stops, with packets still being passed. Anyoen have any clues? Yes there are firewalls between here and there, yes there is NAT at my end...What ports need punching, is rfc2833 the correct settign or should I use inband or info? TIA, I just

Hi, when I put a call on hold from my Nokia E51 (SIP client), the other side does NOT hear music on hold although sip debug / wireshark shows that the E51 tells the asterisk that it now holds the call. Canreinvite is set to "no". Also, when parking a call (features.conf), the parked caller does not hear music on hold. In queues, when using "#" and when using the hold

Hi, I'm playing with * for about 12 years now and since about 10 years, it's my home PBX. I can do pretty much everything I want but one thing I haven't managed yet... Mobile connection via bluetooth... I'm still using a Nokia E51 and the setup and everything works fine. However, on the second or third call, the incoming audio is noise. I have tried alignmentdetection=yes and also

Hello to all. I have installed asterisk-1.6.2.1 + asterisk-addons-1.6.2.0 (for chan_mobile) + bluez-4.60. Bluetooth Dongle: Canyon CN-BTU4 (0a12:0001 Cambridge Silicon Radio, Ltd Bluetooth Dongle (HCI mode)) Device Descriptor: bLength 18